File size: 6,885 Bytes
d3e0df2
3bd5d2b
1f63fcf
 
d3e0df2
ee44eab
1f63fcf
ee44eab
1f63fcf
ee44eab
d3e0df2
ee44eab
 
87505e7
ee44eab
9758654
 
 
 
ee44eab
3bd5d2b
 
ee44eab
00e31d2
ee44eab
 
1f63fcf
 
9758654
24fab16
ee44eab
 
 
 
 
 
 
af74e64
ee44eab
 
af74e64
 
 
 
ee44eab
 
 
 
 
 
 
 
e91d345
ee44eab
 
 
 
 
 
1f63fcf
3bd5d2b
1f63fcf
 
 
 
 
 
 
 
 
 
 
 
 
 
1923ff8
 
 
9758654
1923ff8
 
 
 
1f63fcf
 
d3e0df2
1923ff8
 
 
 
ee44eab
1923ff8
 
 
 
 
ee44eab
1923ff8
 
9758654
ee44eab
1f63fcf
 
 
 
 
 
 
 
 
 
 
 
 
 
 
9758654
1923ff8
1f63fcf
 
9758654
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1f63fcf
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
ee44eab
1f63fcf
 
 
 
9758654
1f63fcf
9758654
 
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
import base64
import os
from functools import partial
from multiprocessing import Pool

import gradio as gr
import numpy as np
import requests
from processing_whisper import WhisperPrePostProcessor
from transformers.models.whisper.tokenization_whisper import TO_LANGUAGE_CODE
from transformers.pipelines.audio_utils import ffmpeg_read


title = "Whisper JAX: The Fastest Whisper API ⚡️"

description = """Whisper JAX is an optimised implementation of the [Whisper model](https://huggingface.co/openai/whisper-large-v2) by OpenAI. It runs on JAX with a TPU v4-8 in the backend. Compared to PyTorch on an A100 GPU, it is over [**70x faster**](https://github.com/sanchit-gandhi/whisper-jax#benchmarks), making it the fastest Whisper API available.

Note that using microphone or audio file requires the audio input to be transferred from the Gradio demo to the TPU, which for large audio files can be slow. We recommend using YouTube where possible, since this directly downloads the audio file to the TPU, skipping the file transfer step.
"""

API_URL = os.getenv("API_URL")
API_URL_FROM_FEATURES = os.getenv("API_URL_FROM_FEATURES")

article = "Whisper large-v2 model by OpenAI. Backend running JAX on a TPU v4-8 through the generous support of the [TRC](https://sites.research.google/trc/about/) programme. Whisper JAX [code](https://github.com/sanchit-gandhi/whisper-jax) and Gradio demo by 🤗 Hugging Face."

language_names = sorted(TO_LANGUAGE_CODE.keys())
CHUNK_LENGTH_S = 30
BATCH_SIZE = 16
NUM_PROC = 16
FILE_LIMIT_MB = 1000


def query(payload):
    response = requests.post(API_URL, json=payload)
    return response.json(), response.status_code


def inference(inputs, language=None, task=None, return_timestamps=False):
    payload = {"inputs": inputs, "task": task, "return_timestamps": return_timestamps}

    # langauge can come as an empty string from the Gradio `None` default, so we handle it separately
    if language:
        payload["language"] = language

    data, status_code = query(payload)

    if status_code == 200:
        text = data["text"]
    else:
        text = data["detail"]

    if return_timestamps:
        timestamps = data["chunks"]
    else:
        timestamps = None

    return text, timestamps


def chunked_query(payload):
    response = requests.post(API_URL_FROM_FEATURES, json=payload)
    return response.json()


def forward(batch, task=None, return_timestamps=False):
    feature_shape = batch["input_features"].shape
    batch["input_features"] = base64.b64encode(batch["input_features"].tobytes()).decode()
    outputs = chunked_query(
        {"batch": batch, "task": task, "return_timestamps": return_timestamps, "feature_shape": feature_shape}
    )
    outputs["tokens"] = np.asarray(outputs["tokens"])
    return outputs


if __name__ == "__main__":
    processor = WhisperPrePostProcessor.from_pretrained("openai/whisper-large-v2")
    pool = Pool(NUM_PROC)

    def transcribe_chunked_audio(inputs, task, return_timestamps):
        file_size_mb = os.stat(inputs).st_size / (1024 * 1024)
        if file_size_mb > FILE_LIMIT_MB:
            return f"ERROR: File size exceeds file size limit. Got file of size {file_size_mb:.2f}MB for a limit of {FILE_LIMIT_MB}MB.", None

        with open(inputs, "rb") as f:
            inputs = f.read()

        inputs = ffmpeg_read(inputs, processor.feature_extractor.sampling_rate)
        inputs = {"array": inputs, "sampling_rate": processor.feature_extractor.sampling_rate}

        dataloader = processor.preprocess_batch(inputs, chunk_length_s=CHUNK_LENGTH_S, batch_size=BATCH_SIZE)

        try:
            model_outputs = pool.map(partial(forward, task=task, return_timestamps=return_timestamps), dataloader)
        except ValueError as err:
            # pre-processor does all the necessary compatibility checks for our audio inputs
            return err, None

        post_processed = processor.postprocess(model_outputs, return_timestamps=return_timestamps)
        timestamps = post_processed.get("chunks")
        return post_processed["text"], timestamps

    def _return_yt_html_embed(yt_url):
        video_id = yt_url.split("?v=")[-1]
        HTML_str = (
            f'<center> <iframe width="500" height="320" src="https://www.youtube.com/embed/{video_id}"> </iframe>'
            " </center>"
        )
        return HTML_str

    def transcribe_youtube(yt_url, task, return_timestamps):
        html_embed_str = _return_yt_html_embed(yt_url)

        text, timestamps = inference(inputs=yt_url, task=task, return_timestamps=return_timestamps)

        return html_embed_str, text, timestamps

    microphone_chunked = gr.Interface(
        fn=transcribe_chunked_audio,
        inputs=[
            gr.inputs.Audio(source="microphone", optional=True, type="filepath"),
            gr.inputs.Radio(["transcribe", "translate"], label="Task", default="transcribe"),
            gr.inputs.Checkbox(default=False, label="Return timestamps"),
        ],
        outputs=[
            gr.outputs.Textbox(label="Transcription"),
            gr.outputs.Textbox(label="Timestamps"),
        ],
        allow_flagging="never",
        title=title,
        description=description,
        article=article,
    )

    audio_chunked = gr.Interface(
        fn=transcribe_chunked_audio,
        inputs=[
            gr.inputs.Audio(source="upload", optional=True, label="Audio file", type="filepath"),
            gr.inputs.Radio(["transcribe", "translate"], label="Task", default="transcribe"),
            gr.inputs.Checkbox(default=False, label="Return timestamps"),
        ],
        outputs=[
            gr.outputs.Textbox(label="Transcription"),
            gr.outputs.Textbox(label="Timestamps"),
        ],
        allow_flagging="never",
        title=title,
        description=description,
        article=article,
    )

    youtube = gr.Interface(
        fn=transcribe_youtube,
        inputs=[
            gr.inputs.Textbox(lines=1, placeholder="Paste the URL to a YouTube video here", label="YouTube URL"),
            gr.inputs.Radio(["transcribe", "translate"], label="Task", default="transcribe"),
            gr.inputs.Checkbox(default=False, label="Return timestamps"),
        ],
        outputs=[
            gr.outputs.HTML(label="Video"),
            gr.outputs.Textbox(label="Transcription"),
            gr.outputs.Textbox(label="Timestamps"),
        ],
        allow_flagging="never",
        title=title,
        examples=[["https://www.youtube.com/watch?v=m8u-18Q0s7I", "transcribe", False]],
        cache_examples=False,
        description=description,
        article=article,
    )

    demo = gr.Blocks()

    with demo:
        gr.TabbedInterface([microphone_chunked, audio_chunked, youtube], ["Transcribe Microphone", "Transcribe Audio File", "Transcribe YouTube"])

    demo.queue(max_size=3)
    demo.launch(show_api=False)